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IIR Filters

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25 views31 pages

IIR Filters

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Paul The Human
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© © All Rights Reserved
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Infinite-duration

Impulse Response
(IIR) Filters

Signal Spectra and Signal Processing


ECE162
Introduction

There are several methods that can be used to design digital filters
having an infinite-duration unit sample response. The techniques
described in this section are all based on converting an analog filter
into a digital filter.
Analog filter design is a mature and well-developed field, so it is not
surprising that we begin the design of a digital filter in the analog
domain and then convert the design into the digital domain.
Analog Filters
An analog linear time-invariant system with system function 𝐻 𝑠 is
stable if all its poles lie in the left half of the s -plane. Consequently,
if the conversion technique is to be effective, it should possess the
following desirable properties:
1. The j-axis in the s-plane should map into the unit circle in the z-
plane. Thus, there will be a direct relationship between the two
frequency variables in the two domains.
2. The left-half plane (LHP) of the s -plane should map into the
inside of the unit circle in the z-plane. Thus, a stable analog filter will
be converted to a stable digital filter.
IIR Filter
A causal and stable IIR filter cannot have linear phase. If the
restriction on physical realizability is removed, it is possible to obtain
a linear-phase IIR filter, at least in principle. However, the signal
processing is computationally cumbersome and appears to offer no
advantages over linear-phase FIR filters. Consequently, when an
application requires a linear-phase filter, it should be an FIR filter.
In the design of IIR filters, we shall specify the desired filter
characteristics for the magnitude response only. This does not mean
that we consider the phase response unimportant. Since the
magnitude and phase characteristics are related, we specify the
desired magnitude characteristics and accept the phase response
that is obtained from the design methodology.
IIR Filter Design by
Approximation of Derivatives
For the derivative dy(t)/dt at time t=nT , we substitute the backward
difference [y(nT ) − y(nT−1)]/T . Thus,

where T represents the sampling interval and y(n) ≡ y(nT ).


IIR Filter Design by Approximation
of Derivatives
Consequently, as shown in the figure, the frequency-domain
equivalent for the relationship is
IIR Filter Design by Approximation of
Derivatives
In the frequency domain, the second derivative 𝑑2 𝑦Τ𝑑𝑡 2 is
equivalent to

It easily follows from the discussion that the substitution for the kth
derivative of y(t) results in the equivalent frequency-domain
relationship
IIR Filter Design by Approximation of
Derivatives
Consequently, the system function for the digital IIR filter obtained
as a result of the approximation of the derivatives by finite
differences is

where Ha(s) is the system function of the analog filter characterized


by the differential equation,
IIR Filter Design by Approximation of
Derivatives
Let us investigate the implications of the mapping from the s-plane
to the z-plane

If we substitute 𝑠 = 𝑗Ω in, we find that


IIR Filter Design by Approximation of
Derivatives
As Ω varies from −∞ to ∞,
the corresponding locus of
points in the z-plane is a
circle of radius 1/2 and with
center at z = 1/2.
Example

Convert the analog bandpass filter with system function

into a digital IIR filter by use of the backward difference for the
derivative.
Solution

Substitute

into H(s). Thus,


Solution

Simplifying,
Solution

The system function 𝐻 𝑧 has the form of a resonator provided that


T is selected small enough (e.g., T ≤ 0.1), in order for the poles to be
near the unit circle. Note that the condition 𝑎12 < 4𝑎2 is satisfied, so
that the poles are complex valued. For example, if T = 0.1, the poles
are located at
Example

Convert the analog bandpass filter

into a digital IIR filter by use of the mapping


Solution

By substituting for s in H(s), we obtain


Solution

We observe that this mapping has introduced two additional poles


in the conversion from 𝐻𝑎 𝑠 to 𝐻 𝑧 . As a consequence, the digital
filter is significantly more complex than the analog filter. This is a
major drawback to the mapping given above.
IIR Filter Design by
Impulse Invariance
In the impulse invariance method, our objective is to design an IIR
filter having a unit sample response ℎ 𝑛 that is the sampled version
of the impulse response of the analog filter. That is,

where T is the sampling interval.


IIR Filter Design by
Impulse Invariance
The frequency response of a lowpass analog filter and the frequency
response of the corresponding digital filter is shown below

Frequency response 𝐻𝑎 Ω of Frequency response of the corresponding


the analog filter digital filter with aliasing
IIR Filter Design by
Impulse Invariance
The digital filter with frequency response H(ω) has the frequency
response characteristics of the corresponding analog filter if the
sampling interval T is selected sufficiently small to completely avoid
or at least minimize the effects of aliasing. It is also clear that the
impulse invariance method is inappropriate for designing high-pass
filters due to the spectrum aliasing that results from the sampling
process.
IIR Filter Design by
Impulse Invariance
To explore further the effect of the impulse invariance design
method on the characteristics of the resulting filter, let us express
the system function of the analog filter in partial-fraction form. On
the assumption that the poles of the analog filter are distinct, we
can write

where {pk} are the poles of the analog filter and {ck} are the
coefficients in the partial-fraction expansion.
IIR Filter Design by
Impulse Invariance
Consequently,
IIR Filter Design by
Impulse Invariance
The system function of the resulting digital IIR filter becomes
IIR Filter Design by
Impulse Invariance
The inner sum converges because 𝑝𝑘 < 0 and yields

Therefore, the system function of the digital filter is

We observe that the digital filter has poles at


IIR Filter Design by
Impulse Invariance
Although the poles are mapped from the s-plane to the z-plane, we
should emphasize that the zeros in the two domains do not satisfy
the same relationship. Therefore, the impulse invariance method
does not correspond to the simple mapping of points.
The development that resulted in H(z) was based on a filter having
distinct poles. It can be generalized to include multiple-order poles.
For brevity, however, we shall not attempt to generalize.
Example

Convert the analog filter with system function

into a digital IIR filter by means of the impulse invariance method.


Solution

We note that the analog filter has


a zero at s= −0.1 and a pair of
complex-conjugate poles at

as illustrated below,
Solution

We do not have to determine the impulse response ℎ𝑎 𝑡 in order to


design the digital IIR filter based on the method of impulse
invariance. Instead, we directly determine 𝐻 𝑧 , from the partial-
fraction expansion of 𝐻 𝑠 . Thus, we have

or
Solution

Since the two poles are complex conjugates, we can combine them
to form a single two-pole filter with system function
Solution
Figure A
The magnitude of the frequency
response characteristic of this filter is
plotted in the figure A for T = 0.1 and
T = 0.5. For purpose of comparison,
we have also plotted the magnitude of
the frequency response of the analog
filter in figure B.
Figure B
We note that aliasing is significantly
more prevalent when T = 0.5 than
when T = 0.1. Also, note the shift of
the resonant frequency as T changes.
𝐸𝑁 =

END

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